A problem that exists with currently deployed public pay phones is that these devices do not generate money for the owner/operator when the phones are not in use. To make up this shortcoming, an owner of a public pay phone will try and place a quantity of these devices in high traffic areas where the percentage of use in any time period will be higher than in places where large numbers of people do not go. Additionally, the owner may attempt to recover expenses by charging higher than normal access and per period of time fees to make up for the less frequent use of the phone.
Many existing public pay phones are connected to the analog telephone network, otherwise known as Plain Old Telephone Service (POTS), and thus are limited in the kinds of service that can be provided to a public customer. For example, compared to Voice over IP (VoIP) phones, analog public phones can provide many of the business level (CENTREX) functions that VoIP phones can provide, but they cannot provide high speed data services, such as video, given that POTS line connections are limited to 64 kilobits per second (kbs) in either direction. VoIP phones, on the other hand, can be easily connected to network connections which provide much higher rates of data throughput, thus supporting services like video over that connection.
Not all VoIP phones and their connections are capable of supporting video. The reasons are many, but can be sufficiently discussed with a few items: A) protocols B) bandwidth C) device hardware support and D) quality of service of both the network connection and the VoIP phone itself.
Hardware support for video is the most important feature for a VoIP device to be able to support any VoIP and/or video stream. VoIP phones and devices tend to fall into one of four classes: A) Analog Telephone Adapters (ATA's) B) Audio only VoIP phones, C) Video VoIP phones and D) VoIP soft phones. ATA's provide an analog telephone connection to a POTS phone, while transforming the signal on the other end to a digital packet protocol that is compatible with the VoIP Service Provider's (VSP) network that a customer has engaged to make calls over a digital network, including the Internet, terminating to another POTS or VoIP phone or other communications device. By definition, ATA's will not support video streaming, as POTS phones do not. Additionally, ATA's do not come with any hardware that would support a video stream, should it receive one. This is done primarily as a cost savings measure, in addition to the fact that with POTS phones it is an unneeded feature. These devices typically are priced very low compared to many dedicated, full featured, VoIP phones, so as to provide the least expensive way for a customer to enjoy the benefits offered by a VSP.
Audio only VoIP phones behave and act like POTS phones with the same class of features. The difference lies in the connection. Audio only VoIP phones require a digital broadband connection to make and receive telephone calls. Audio only VoIP phones do have an advantage over their POTS counterparts in that they can utilize multiple protocols to send and receive audio in order to save bandwidth, or transmit a higher quality signal to the receiving end.
Popular audio coder/decoders (codecs) range from G.729A/B at the low end of the bandwidth scale, along with G.726, G.723 to G.711 at the high end of bandwidth usage. This flexibility in bandwidth requirements allows Audio only VoIP phones to service many different types of needs with many different types of broadband connections.
G.729 is mostly used in VoIP applications for its low bandwidth requirement. Standard G.729 operates at 8 kbit/s, but there are extensions, which provide also 6.4 kbit/s and 11.8 kbit/s rates for marginally worse and better speech quality respectively.
Video VoIP phones are the most sophisticated and generally the most expensive of the VoIP phone hardware family. Much more processing is required of them to produce both audio and video signaling, as well as telephone service features. In addition to the audio codec pantheon supported by Audio only VoIP phones, Video VoIP phones add video codecs, such as H.261, H.263 and H.264 to the mix. Many implementations of Video VoIP phones include computer like functionality, such as browsers, network utility and configuration applications, and aesthetic applications to assist users in configuring, managing and adapting the phone to their environment and needs. Computing power in Video VoIP phones comes near to and sometimes rivals that found in many laptop and desktop computers. This processing capability is what makes the additional functionality and video processing in real time possible.
While it may appear rather strange to discuss VoIP soft phones in the same category as hardware versions of the same, there is a high degree of similarity with the hardware version. Essentially, VoIP soft phones depend upon a host computer and operating system to provide the environment where it can operate. Depending on the computing power and capacity of the host machine, a soft phone can make regular audio only calls, video calls, instant messaging, file transfers, e-mail, conference calls, and many other “computer and phone like” functions. The positive to this hardware/software relationship, is the soft phone can be configured or expanded to take advantage of the additional processing, storage, and resource capacity of the host machine. Also, because the soft phone runs on general purpose hardware, there are many opportunities to apply this combination in solving many telecommunication problems that would not be satisfied sufficiently with a dedicated phone function and styled device. The downside to soft phone embodiment entails costs (much more costly than dedicated VoIP hardware), physical size, power requirements, and lack of portability (in some cases). This also limits in some respects the telecommunications problem set that can be addressed efficiently.
Because of the aforementioned flexibility of the VoIP soft phone, it is possible to address, more effectively, the main business limitations experienced by currently deployed public pay phones and kiosks. However, there are some basic requirements that must be met, given current technology, in order for the VoIP soft phone to function as a viable alternative to existing public pay phones and kiosks.
For starters, a broadband connection of sufficient speed and quality must connect the VoIP soft phone to the VSP that is providing the service. Standard POTS lines limit their frequency response to 4 KHz, and their digital counterpart (DSO), limits its clocking to 64 kbps for connections of any type. This is fast enough to accurately render an audio signal with a bandwidth of 200 Hz to 3600 Hz, which has been used for many years to carry phone conversations on POTS analog lines. In the VoIP world, several popular audio codecs can operate within this bandwidth range: G.711 requires 64 Kbs to render an audio signal, G.723 uses between 16 Kbs and 40 Kbps, and G.729 uses between 4 Kbps and 16 Kbps.
Digital Video signals in the VoIP world come in a variety of quality and physically rendered image sizes to meet a number of different needs. In general, the bandwidth requirements for a video VoIP signal can range from 40 Kbs to 2 Mbs. The higher the bandwidth, the bigger the image size and the better the quality of that video. At the low end of the bandwidth scale image resolutions can range from 88×72 pixels, to 178×144 pixels with frames rates of about 5+ frames per second. At 2 Mbs, the image resolutions approach television quality high definition rendering, with a size of 704×576 pixels (and larger in some cases) with frames rates up to 30 frames per second.
While it is desirable to have as high a quality of video image and audio signal, that desire must be balanced against financial costs associated with the generation and transport of that signal. Current popular video codecs include H.261, H.263 and its variants, and the most recent codec, H.264. H.261 is the oldest of the codecs, introduced in 1990, and requires the most bandwidth of the three mentioned to produce the equivalent video resolution and frame rate. H.261's was the first practical digital video standard and set the stage for the creation of subsequent standards such as MPEG-1, MPEG-2 (H.262) and so on. H.263 was introduced in 1995/1996 and provided better image compression and bandwidth usage than its older sibling. This codec family is still in widespread use today in the VoIP world and provides a good quality image for its intended design use in video conferencing. It is efficient enough that Adobe® adopted the codec for use in their very popular Flash 8 format. In May 2003, H.264 (also known as MPEG-4) was introduced as the most efficient codec for compressing and delivering video signals over low bandwidth connections. It was also the first of these codecs to be designed with the intent to operate efficiently on RTP/IP packet networks and multimedia telephony networks. Using this codec makes Video VoIP telephony even more cost effective.
A reasonable comprise that is “good enough” for a typical video phone conversation, would require the use of either H.263 or H.264 for video at Common Intermediate Format (CIF) resolution of 356×288, coupled with one of the following audio codecs: G.729, G.726, or G.723. Depending on the combination of codecs, hosting hardware, and VSP, bandwidth requirements could run from about 150 Kbs to 256 Kbs to support video calls at up to 30 fps.
A final requirement to bring VoIP service in line with existing POTS service centers on Quality of Service (QoS). QoS, in essence, is the practice of managing the streaming packets that comprise a VoIP phone call so that as many packets as possible arrive between sender and receiver in a timely manner and in the proper order. Since POTS is analog in nature, there is no need to manage anything other than signaling, as all call information (including control and voice) arrive at their intended destination on time and always “in order”. There are no packets to manage. VoIP on the other hand is built on top of the TCP/IP protocol, which itself is physically transported across copper and optical lines as Ethernet or ATM packets. By design, TCP/IP is “store and forward”, meaning that packets generated between two end points do not have to arrive at a particular time, nor in order. Because TCP/IP is not “real time”, nor packet ordered in its delivery, this gives the protocol and applications that use it for data transfers over that connection, a great deal of flexibility on how they transmit and route data from one point to another. This flexibility is not conducive to real time applications such as telephony, which requires deterministic packet order and delivery times for proper telephone call packet conversation to analog signaling at both ends of the phone call. To compensate for this feature of TCP/IP, many VoIP devices, software and VSP's use techniques such as packet ordering, jitter buffers, and packet prioritization to ensure QoS.
What is needed is a public conveyance that couples analog and digital telephony, video, and additional related services. What is also needed is a method of near if not continuous income generation to provide these services at commodity pricing to encourage increased use, as well as a more stable income to the owner of this conveyance. The present invention solves these problems with the use of scheduled paid advertising and other income generating business techniques, digital communications protocols, broadband connections, video hardware support, and QoS techniques to create a novel delivery of information in telephony and related services to the public.